Fix code style and gyp files in cast to build cast_unittest

media/cast/rtp_receiver/receiver_stats_unittest.cc

 // Copyright 2013 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #include <gtest/gtest.h>
 
 #include "base/test/simple_test_tick_clock.h"
 #include "base/time/time.h"
 #include "media/cast/rtp_common/rtp_defines.h"
 #include "media/cast/rtp_receiver/receiver_stats.h"
@@ skipping 12 matching lines @@
         rtp_header_(),
         fraction_lost_(0),
         cumulative_lost_(0),
         extended_high_sequence_number_(0),
         jitter_(0) {
     testing_clock_.Advance(
         base::TimeDelta::FromMilliseconds(kStartMillisecond));
     start_time_ =  testing_clock_.NowTicks();
     delta_increments_ = base::TimeDelta::FromMilliseconds(kStdTimeIncrementMs);
   }
-  ~ReceiverStatsTest() {}
+  virtual ~ReceiverStatsTest() {}
 
   virtual void SetUp() {
     rtp_header_.webrtc.header.sequenceNumber = 0;
     rtp_header_.webrtc.header.timestamp = 0;
     rtp_header_.webrtc.header.ssrc = kSsrc;
   }
 
   uint32 ExpectedJitter(uint32 const_interval, int num_packets) {
     float jitter = 0;
     // Assume timestamps have a constant kStdTimeIncrementMs interval.
@@ skipping 104 matching lines @@
   EXPECT_FALSE(fraction_lost_);
   EXPECT_FALSE(cumulative_lost_);
   // Build extended sequence number (one wrap cycle).
   uint32 extended_seq_num = rtp_header_.webrtc.header.sequenceNumber - 1;
   EXPECT_EQ(extended_seq_num, extended_high_sequence_number_);
   EXPECT_EQ(ExpectedJitter(kStdTimeIncrementMs, 300), jitter_);
 }
 
 }  // namespace cast
 }  // namespace media